Lead Engineer

Overview

Lead Engineer to own the real-time voice and telephony platform that powers live AI conversations, setting technical direction from the carrier and SIP trunk through to the AI agent. This is a hands-on leadership role, with around 50–70% of time spent designing and building and the rest leading and mentoring a team of 3–6 engineers.

What you'll do

• Own the end-to-end architecture of the voice and telephony stack, covering SIP signalling, media servers, WebRTC and call routing from the carrier to the AI agent.
• Design and run real-time media pipelines (FreeSWITCH, Asterisk, Kamailio or LiveKit) that stream audio to ASR, LLM and TTS within tight end-to-end latency budgets.
• Lead carrier and telephony-provider integrations: SIP trunks, DIDs, number provisioning and multi-provider failover.
• Drive call quality and reliability across codecs, jitter, packet loss, echo and barge-in.
• Define and own SLOs for latency, uptime and call success rate.
• Scale the platform for high call concurrency through capacity planning, autoscaling of media workloads on Kubernetes and cost per call minute.
• Set the engineering bar for the team: lead system design, review code and designs, make build-vs-buy calls and manage tech debt.
• Lead and mentor a team of 3–6 engineers: plan and break down work, grow engineers through feedback and take part in hiring.
• Own production health for live call traffic, including observability (Datadog, Sentry), on-call, incident response and postmortems.
• Partner with product and AI teams on the platform's multi-agent orchestration and automation layer.
• Work closely with the ML team to integrate in-house models, drive benchmarking and evaluation, and help build the next generation of in-house AI models.

What you'll need

• 7–10 years of backend development experience in Python or Java, including time spent building high-scale, low-latency real-time systems.
• Proven experience as a tech lead: guiding a team through design and delivery, mentoring engineers and staying hands-on in the codebase while leading a team of 3–6 engineers.
• Deep, hands-on expertise in SIP, RTP/SRTP, SDP and WebRTC.
• Has built or operated media and signalling stacks such as FreeSWITCH, Asterisk, Kamailio/OpenSIPS or LiveKit in production.
• Experience with carrier integrations and SIP trunking, plus CPaaS providers such as Twilio, Exotel or Plivo.
• Strong grasp of audio codecs and NAT traversal, and debugging call quality using tools such as Wireshark or sngrep.
• Experience shipping Voice AI in production: streaming ASR/TTS, real-time inference and end-to-end latency tuning.
• Experience integrating LLMs, vector DBs or AI agent frameworks into production systems.
• Strong command of system design, concurrency and distributed systems, with a track record of owning architecture decisions.
• Experience with cloud infrastructure, databases, and message queues, plus containerized deployments with Kubernetes and Docker.
• Eligible to work in India.

Nice to have

• Prior work leading AI-driven or telephony-driven platforms at scale.
• Experience building contact-centre, IVR or outbound dialer systems.
• Experience collaborating with ML/research teams on model benchmarking, evaluation or observability.
• Contributions to open-source telephony or WebRTC projects.

Details

• Location: Bengaluru, India.